Rajarshi Shahu College, Pune

Department of Information Technology

Presentation on

By Ms. Shriya Arvind Jadhav Ms. Nikheta Dattatray Godse Ms. Sarika Vidhyadhar Gawade

Guide Prof. Mrs. D.H.Patil

Contents
Introduction of system Literature Survey System Architecture System Description System Requirements Modules Descriptions Project Flow Conclusions

Introduction
Providing mobile environment for VoIP calls You can use SIP phones to call your friends and family at any geographical location Provide low rate calls or even free calls.

Hello..

Hi !

Literature Survey
For developing this project we have referred number of IEEE papers for gaining better knowledge in this subject. Some of those papers are as follows:

Call set-up time modeling for SIP-based stateless and stateful calls in Next Generation Networks

Voice over Internet Protocol on Mobile Devices

Efficient User Controlled Inter-Domain SIP Mobility Authentication, Registration, and Call Routing

SYSTEM ARCHITECTURE
SIP PROXY

Registration
100.101.102.103

Registration
100.101.102.103

Signaling Signaling Media

SIP User Agent
.

SIP User Agent

What is IP Telephony?
Internet Telephony (Voice over IP) essentially means a voice message transmitted using the Internet Protocol Basically this means sending voice information in digital form in discrete packets as opposed to the traditional circuit-committed protocols of the public switched telephone network.

PBX: Private Branch Exchange is a telephone system within an enterprise that switches calls between enterprise users

IP Telephony v/s PSTN

PSTN : Static Switching
Telephone central office

Calling

Call connected

IP Telephony : Dynamic Routing

SERVER Calling Internet

Call connected

What is SIP and Why SIP?

In the world of VoIP, SIP is a call setup protocol that operates at the application layer

SIP is the core protocol for initiating, managing and terminating sessions in the Internet These sessions may be text, voice, video or a combination of these SIP sessions involve one or more participants and can use unicast or multicast communication

SIP Entities
User Agent User Agent Client (UAC)a client application that initiates SIP requests. User Agent Server (UAS)a server application that contacts the user when a SIP request is received and that returns a response on behalf of the user. Proxy Server Redirect Server Registrar

SIP Message
There are two types of SIP messages: 1]Request methods REGISTER : Used by a client to register with a SIP server INVITE : Initiates a call ACK : Confirms a final response for INVITE CANCEL : Used to cancel a pending request BYE : Terminates a call OPTIONS : Used to query a server 2]Response code 1xx : searching, ringing 2xx : success 3xx : forwarding 4xx : client mistakes 5xx : server failures 6xx : busy, refuse, not available anywhere

Architecture Diagram

SIP Flows - Basic

User A INVITE User B

180 : Ringing
200 OK : Answers the call ACK Talking Talking

200 OK

SIP Flows Via Proxy

The SIP proxy only participates in the SIP user authentication and messages, once the call is set up, the phones send their voice traffic directly to each other without involving the proxy

User A INVITE

Proxy

MIT.EDU

User B INVITE

Trying
180- - Ringing

180 - Ringing 200 OK : Answers the call

200 - OK ACK Talking Hangs up

Talking 200 - OK

System Requirements
1) S/W Requirements Operating System Microsoft Windows XP. miniSipServer 2) Programming Languages Dot Net 3) Hardware Requirements Processor: P4/Dual core RAM: 64MB Hard Disk: 10 GB

Main Module
Basic Two Modules
Server Side - miniSIPserver for routing calls

Client Side - For making calls and receiving calls

Server Settings
The software works as a fully featured telephone switch connecting to phone lines and extensions using state-of-the-art VoIP technology. Offering all the normal features of a traditional PBX Works with Any Standard Soft Phone Support static as well as dynamic IP

Add new users

The procedure of adding new user to the network is not a tedious job from this application. Thus provides, Easy adding user Easy to modify the current user Easy deletion of existing user

Project Flow
STEP 1]

SIP account setting

USERNAME

PASSWORD
DOMAIN NAME SERVER IP

Local User Status

At the server side, it shows the status of the registered users. Here the BLUE color icon shows the active users i.e. users that are online at that instant.

Step 2]

For caller :
Dial user number Make call Hang-up the call after connection done

Line status :
Line1- Dialing /Trying, for establishing the call Line1-Ringing,indicating call got established Line1- Connected, indicates that the call got connected with end user Line1-Busy,line is busy with other end user on call Line2-Free,this line can be use to make or accept call.
.

Step 3]

For callee: When the caller makes a call, on the callee side the callee gets this window having the following options: Accept the call Reject call

Advantages and Disadvantages

1) Advantage Reduce long distance calling cost. Free calls for SIP to SIP user agents. Portable Clear voice quality 2) Disadvantage This application is expensive to use No facility to make Emergency Calls

Applications
Setting up a Private LAN in office Voice Mail Service Ringing Group Service In VPN help to make call within the network

Future Scope
Compatibility with any mobile phones Developing more APIs using SIP like Virtual conference room service Voice message IP Billing

Conclusion.
Thus this application will help people to communicate staying in any Geographical location at low cost.

References
Technical Papers. [1] Pirhadi, M. Hemami, S.M.S. Tabrizipoor, A.I. Sci. & Res. Branch, Islamic Azad Univ., TehranAdvanced Communication Technology, in Proc. 11th IEEE Workshop on Applications of IP Telephony, Washington DC pp,2008,Page(s): 34-34 [2] Khoury, J.S. Jerez, H.N. Abdallah, C.T. Univ. of New Mexico, Albuquerque Mobile and Ubiquitous Systems: Networking & Services in Proc. 4th IEEE Workshop on Applications of Networking, Washington DC pp,2005,Pages 333-333 Books. 1. Session Initiation Protocol-Controlling Convergent Networks by Travis Russell 2. Guide to Telecommunications Technology by Tamara Dean 3. Internet Communications Using SIP: Delivering VoIP and Multimedia Services with Session Initiation Protocol (Networking Council) by Henry Sinnreich and Alan B. Johnston 4. C# Programming Languages by Anders Hejlsberg Mads Torgersen, Scott Wiltamuth, and Peter Golde. Some online links. www.google.com www.sipcenter.com www.myvoipapp.com/minisipserver

Thank You